FIG. 1 shows a simplified block diagram of the low frequency part of the transmission path for a speech signal in a radiotelephone according to prior art. The transmission path consists of a microphone 1, a microphone amplifier 2, a band-pass filter 3, a compressor 4, a pre-emphasis amplifier 5, a soft clipper 6, a hard clipper 7, a low-pass/band-stop filter 8, and an adder 9.
The microphone 1 transforms an acoustic sound signal into an electrical signal. The microphone amplifier 2 amplifies the low level sound signal. The band-pass filter 3 limits the bandwidth of the transmitting signal. The pass-band of the filter 3 is from about 0,3 kHz to 3 kHz.
The compressor 4 compresses the dynamics of the transmitting signal with a compression ratio of 2:1. When the signal has a certain level the gain of the compressor 4 is 0 dB. This level is called the ineffective level. If the level of the input signal is +x dB compared with the ineffective level, then the level of the output signal is +x/2 dB. If the level of the input signal is -x dB compared with the ineffective level, then the level of the output signal is -x/2 dB. The gain of the compressor 4 settles slowly. The rate of change of the gain is characterized by the strike time and the return time. The pre-emphasis amplifier 5 emphasizes frequency components at the higher end of the frequency band more than the frequency components at the lower end of the transmitting frequency band. The pre-emphasis is +6 dB/octave.
The soft clipper 6 and the hard clipper 7 limit the measured value normally the peak to peak value of the amplitude of the transmitting signal, so that the deviation on the radio path will be smaller than the limit value required by the radio telephone system in question. In the prior art device shown in FIG. 1 the the soft clipper measures only its own output signal.
The gain of the soft clipper 6 is a constant G (e.g. 0 dB), when the measured peak to peak value of the input signal is lower than a selected VINpp, max. When the measured peak to peak value of the input signal exceeds the value VINpp, max the gain of the soft clipper 6 decreases so that the measured peak to peak value of the output signal is kept at a constant value G * VINpp, max regardless of the input signal. The gain of the soft clipper 6 will settle slowly. The rate of change of the gain is characterized by the stroke time and the return time.
In order to prevent transient deviation overshots the soft clipper 6 is followed by a hard clipper 7. When the instantaneous value VIN of the signal at the hard clipper 7 input is lower than VIPmax and higher than VINmax (VIPmax=-VINmax), then the value of the output signal is higher than VIPmax, then the value of the output signal is the gain G of the hard clipper 7 multiplied by VIPmax (G * VIPmax). When the instantaneous value of the signal is lower than VINmax, then the value of the output signal is the gain G of the hard clipper 7 multiplied by VINmax (G * VINmax).
The low-pass/band-stop filter 8 limits the bandwidth of the transmitting signal (attenuates any generated harmonic frequency components) and attenuates noise coming from the speech path to the control signal bandwidth. The adder 9 adds the speech signal, the data signal DATA and the control signal SVS. The sum signal MOD is the signal modulating the modulator.
FIG. 2 shows an embodiment of the soft clipper 6 according to prior art. The shown embodiment is very well suited to be integrated in CMOS technology. The soft clipper 6 consists of a digitally controlled controllable amplifier 10, a window comparator 11 and of control logic 12.
The input signal CTRL has a level corresponding to the measured peak to peak value of the signal. The input signal IN is the input signal to the soft clipper 6 and the output signal OUT is the output signal of the soft clipper 6. The signal to be measured in the block diagram of FIG. 1 is the output signal of the soft clipper 6. The DC references REFP and REFN determine the maximum value of the measured peak to peak value of the output signal.
The window comparator 11 compares whether the monitored signal CTRL is within the references or outside them. The logic 12 reads the comparison result at the rising or falling edge of the clock signal CLK.
The logic 12 contains a control word of L bits, of which the N highest bits control the controllable amplifier 10. The core of the logic 12 comprises a counter of L bits. In an alternative embodiment the counter is replaced by a simple arithmetic unit counting in serial mode. The higher the value of the word formed by the N most significant bits the lower the gain of the controllable amplifier 10.
According to the value of the comparison result from the comparator 11, either a constant A (Attack) is added to the control word of L bits, or a constant D (Decay) is subtracted from it. Counting overflow is prevented; the minimum value of the control word is 0 and its maximum value is Lmax (2.sup.N -1). The constant A is large compared with the constant D.
When the comparison value of the comparator 11 is outside its limits the constant A is added to the control word of L bits. The strike time of the soft clipper 6 is determined by the constant A. When the comparison value of the comparator 11 is within its limits the constant D is subtracted from the control word of L bits. The return time of the soft clipper 6 is determined by the constant D.
If the input signal level to soft clipper 6 increases suddenly, the soft clipper slowly starts to limit its output signal or the input signal level to hard clipper 7. The hard clipper 7 cuts off the signal peaks and thereby forms harmonic components. Harmonic components also exist in a continuous state, if the signal is cut off already before the soft clipper 6.
The low-pass/band-stop filter 8 after the hard clipper 7 typically has a considerable delay distortion. A compensation of the delay distortion with a delay equalizer would complicate the transmission path and would increase the noise, so in the design of this filter 8 no attention is paid to the phase response but only to the amplitude response.
Due to the delay distortion of the low-pass/band-stop filter 8 the signal contains harmonic components and in the worst case it resembles a square wave and will be distorted. The measured peak to peak value of the output signal does not equal the gain of the filter 8 multiplied by the measured peak to peak value of the input signal, but is C multiplied by the gain of the filter 8 multiplied by the measured peak to peak value of the input signal, where C is a function of the frequency and typically considerably greater than 1, especially at frequencies in the lower end of the speech band, e.g. 1,4.
Generally the transient deviation overshots do not pose any problems, but prolonged overshots may do. The problems most easily detected are the interruptions of calls.